Beginning setup question re: SIP and Extensions


#1

I’m currently setting up my first Asterisk system (1.8.7.0) with Debian and was having issues registering my SIP Softphones which are X Lite 4 all on my local LAN. I’m currently able to dial x1000 from my softphone and it plays what sounds like from the documentation Ive read the Hello-World File. When I browse both the SIP and Extensions.conf files located in the /etc/asterisk directory I do not see anything referencing my SIP Softphone- I have not edited these files since initial load and they are at their defualts? Ive also logged into the Debian Gui and noticed that I also have another Asterisk Directory located in the /Usr/Src/Asterisk-1.8.7.0/Configs/ which has a SIP and Extensions.Conf file in it as well- This directory seems to also have all of the sample files I loaded at beginning of install.

SIP SoftPhone settings:

ID: 1001 * This SIP softphone can dial 1000 to reach the Hello-World prompt
Domain: IP address of server
Password: 1001
authorization name: 1001

ID: 10002 * All the same settings on soft phone but gets No matching peer found Notice 2921 when trying to register
Domain: IP address of same server
Password: 1002
authorization name: 1002

SIP Show Peers returns no results of registered SIP Phones

I’ve reloaded the dialplan and restarted Asterisk- Im just trying to understand how Asterisk is dialing extension 1000 when there are no SIP Peers registered and I do not see any configurations in place for the extension to call that voice file. Is there another location that has this information that im not checking?

Thank you in advance for any input


#2

This path comes from Your installation sources and inccludes the sample-conf-Files from Asterisk, they aren’t used by any running Asterisk-instance (except one wishes to manipulate /etc/asterisk/asterisk.conf or starts asterisk with -c /Usr/Src/Asterisk-1.8.7.0/Configs :smiley: )
So forget these files, they are just examples for reference …

As You run Your asterisk with just the sample config-files copied to /etc/asterisk You’re also running a example dialplan allowing any phone from any IP to dial some extensions. You may check the reachable extensions by issuing

from the Asterisk-CLI. In the basic configuration which is running at Your side so called guestaccess is permitted for SIP as well as for IAX2-Clients. The access is granted to the context default and it’s inclusions. That’s why it’s possible to dial extension 1000 without registering the SIP-Client itself (as there is no definition of a peer 1000 in the sip.conf.example You’re using at the moment).
And this is also the reason, why You could for example make no call to extension 1001 or 1002 as they aren’t registered with asterisk and aren’t extensions where asterisk would start any application inside the sample dialplan.

You should however start to read the Asterisk-Book to get a impression of the interaction between the several config-files and the basic concepts of asterisk.


#3

The sample sip.conf has allowguest=yes, which is normally not desirable on a production system. It allows anyone to call into the default context. It is set unsafe on the sample system precisely to let people get the sample system up with as little configuration work as possible.

Calls come from devices, not extensions, and there doesn’t need to be anything in extensions.conf corresponding to a device for incoming calls to work (although there must be something, as every incoming call has an outgoing side).


#4

Thank you guys for responding so quickly. I’m definetely going through the Asterisk book as well as all the forums here and other tutorials online but these were just a few things that I couldnt piece together. Thank you for helping me clarify the distincitions between the directories and why my softphone has functionality without additional configurations i expected I had to input.

Many Thanks again!